The invention concerns a method for time-synchronous data transfer, particularly of voice and video messages, over a network, particularly the Internet, between at least two terminals where a connection between the terminals is set up using the SIP protocol and at least one SIP server.
Methods for transfer of time-synchronous data, such as voice data, over networks, particularly over the Internet, are gaining importance, because for private users as well as for enterprises cost saving on telephone calls are possible. For conducting telephone calls between two terminals over a network, particularly using the Internet protocol, means for signalling call set-up and tear-down are required. SIP, the Session Initiation Protocol is one of the protocols used for this purpose. It was standardized by the IETF, the Internet Engineering Task Force.
A caller may send a SIP message for setting up a call by using his/her terminal. The message notifies the callee that the caller intends to set up a call. The terminal of the callee would then for example ring and notify the terminal of the caller by another SIP message that ringing has started. If the callee operates her/his terminal such that it accepts the call, then the terminal sends another SIP message to the terminal of the caller for notifying it that now transmission of time-synchronous data, for example voice or video data, can start. The SIP protocol is also used for signalling tear-down of a connection.
Establishing a concrete voice connection and coding and sending time-synchronous voice data is not supported by SIP. For this, the terminals communicate with each other, for example, by negotiating about the kind of connection or data transfer to use and a coding method for voice data. SIP supports establishing a connection insofar, as it includes an exchange of terminal properties. This includes the kinds of voice coding that the terminals support, the addresses of the terminals to which voice traffic is to be sent, and some other terminal specific properties.
Another functionality of SIP is finding a callee at his/her current location. The first message from a caller to a callee if typically not sent directly to the callee's terminal, but to a SIP server, which is usually configured as SIP proxy server. At this server, a company XYZ provides an address sip://customer@xyz.de to one of its customers.
Now, the customer can register her/his current terminal at the SIP proxy server provided by company XYZ. His current terminal might be his work phone, his home phone, his mobile phone, or any other SIP-enabled phone. The terminal of the caller then sends the first message to sip://customer@xyz.de. There the proxy server forwards this message to the terminal that the customer has registered. The SIP server would also forward the reply of customer's terminal in the opposite direction.
As for the conventional technique, the following references are known.
Reference for Differentiated Services:                “RFC 2475 An Architecture for Differentiated Service,” S. Blake, D. Black, M. Carlson, E. Davis, Z. Wang, W. Weiss, December 1998 (Format: TXT=94786 bytes) (Updated by RFC3260) (Status: INFORMATIONAL)        
Reference for RSVP:                “RFC 2205 Resource ReSerVation Protocol (RSVP), Version 1 Functional Specification,” R. Braden, Ed., L. Zhang, S. Berson, S. Herzog, S. Jamin, September 1997 (Format: TXT=223974 bytes) (Updated by RFC2750) (Status: PROPOSES STANDARD);        “RFC 2210 The Use of RSVP with IETF Integrated Services,” J. Wroclawski, September 1997 (Format: TXT=77613 bytes) (Status: PROPOSED STANDARD)        
Reference for SIP:                “RFC 3261 SIP: Session Initiation Protocol,” J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, June 2002 (Format: TXT=647976 bytes) (Obsoletes RFC 2543) (Updated by RFC 3265) (Status: PROPOSED STANDARD).        
For the known methods of transferring time-synchronous data over the basic Internet, there is the particular problem that they give no guarantees for Quality of Service (QoS), for example available bandwidth and packet delay of a connection. This may lead to bad quality of voice transmission, because packets containing the coded voice arrive in an order different to the one they were sent in, because packets are damaged or dropped during transfer, or because packets are transferred with high delay. For time-synchronous data, such as voice or video, this leads to bad QoS. The lack of QoS is one of the reasons for the limited acceptance of time-synchronous services, particularly Internet telephony, so far. Also SIP does not have any built-in mechanism to support Quality of Service (QoS) to the time-synchronous data transfer it signals.
Enhancements of the basic Internet, such as Integrated Services and Differentiated Services support QoS for Internet connections, but it requires additional signalling and network management functions. Integrating SIP signalled IP telephony or video transfer with these methods for QoS provisioning would be a significant technology improvement and it would increase the acceptance of Internet telephony, but its available is very limited, so far.
Existing suggestions on how to perform this integration are based on the idea that the telephony terminals themselves try to reserve resources for their calls by using other means of signalling, independent of SIP. An example is the Resource reSerVation Protocol (RSVP) of Integrated Services. However, this approaches do not scale sufficiently with an increasing number of users, such that existing QoS provisioning systems cannot deal with a high rate of reservation requests.